lirees Posted February 4, 2011 Report Share Posted February 4, 2011 i would connect two offices through a vpn connection, but I have many problems i have create two trunk gateway in this way: office1 ( 172.16.10.210 ) Name: office2 Type: sip gateway Direction: in and out Trunk Destination: generic sip server State: enabled Account: 123 Domain: 192.168.1.60 Username: 123 Password: **** Proxy Address: 192.168.1.60 office2 ( 192.168.1.60 ) Name: office1 Type: sip gateway Direction: in and out Trunk Destination: generic sip server State: enabled Account: 123 Domain: 172.16.10.210 Username: 123 Password: **** Proxy Address: 172.16.10.210 the extension in the office1 is 2xx and in the office2 is 3xx the dial plan for office1 is : pref 100 Trunk office2 Pattern: 3xx Replacement: * the dial plan for office2 is : pref 100 Trunk office1 Pattern: 2xx Replacement: * when i make a call from office1 to office2 and viceversa i give this error : [5] 2011/02/04 11:35:48: SIP Rx udp:192.168.1.50:5060: INVITE sip:200@172.16.10.210;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-b92a3ef5cc37de1cd2c5a3beef75589d;rport From: "poa" <sip:300@192.168.1.50;user=phone>;tag=18523 To: <sip:200@172.16.10.210;user=phone> Call-ID: 9e1eef98@pbx CSeq: 12399 INVITE Max-Forwards: 70 Contact: <sip:123@192.168.1.50:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 P-Asserted-Identity: "vm" <sip:123@172.16.10.210> Content-Type: application/sdp Content-Length: 327 v=0 o=- 17786 17786 IN IP4 192.168.1.50 s=- c=IN IP4 192.168.1.50 t=0 0 m=audio 55380 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/02/04 11:35:48: Last message repeated 2 times [5] 2011/02/04 11:35:48: SIP Tx udp:192.168.1.50:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-b92a3ef5cc37de1cd2c5a3beef75589d;rport=5060 From: "poa" <sip:300@192.168.1.50;user=phone>;tag=18523 To: <sip:200@172.16.10.210;user=phone>;tag=d2057dfc98 Call-ID: 9e1eef98@pbx CSeq: 12399 INVITE Content-Length: 0 [5] 2011/02/04 11:35:48: Received incoming call without trunk information and user has not been found [5] 2011/02/04 11:35:48: SIP Tx udp:192.168.1.50:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-b92a3ef5cc37de1cd2c5a3beef75589d;rport=5060 From: "poa" <sip:300@192.168.1.50;user=phone>;tag=18523 To: <sip:200@172.16.10.210;user=phone>;tag=d2057dfc98 Call-ID: 9e1eef98@pbx CSeq: 12399 INVITE Contact: <sip:200@172.16.10.210:5060> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2011/02/04 11:35:48: SIP Rx udp:192.168.1.50:5060: ACK sip:200@172.16.10.210;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-b92a3ef5cc37de1cd2c5a3beef75589d;rport From: "poa" <sip:300@192.168.1.50;user=phone>;tag=18523 To: <sip:200@172.16.10.210;user=phone>;tag=d2057dfc98 Call-ID: 9e1eef98@pbx CSeq: 12399 ACK Max-Forwards: 70 Contact: <sip:123@192.168.1.50:5060;transport=udp> P-Asserted-Identity: "vm" <sip:123@172.16.10.210> Content-Length: 0 i have not found any document about connect two office through a vpn connection thanks Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 4, 2011 Report Share Posted February 4, 2011 The PBX does not care if it is VPN, public Internet, private addresses or whatever. The key point is if addresses are routable and from the log above that seems to be the case (no problem). In the example above, you say the PBX is on 192.168.1.60, but the packet is received from 192.168.1.50. Thats why the PBX cannot match the incoming call to a trunk. Simple typo? Quote Link to comment Share on other sites More sharing options...
lirees Posted February 4, 2011 Author Report Share Posted February 4, 2011 is not a typo error, you're right, the ip of the office2 is 192.168.1.50 i have configure the trunk with the wrong ip . now i call the extension without problem but if i try to call a external numer form the exstension of office2 through the line of the office1 i give : 404 Not Found could be a problem of the dial plan ?? DP office1 pref 70 trunk office2 Pattern 3xx Replacement * pref 100 trunk voip Pattern * Replacement * this is the log : [5] 2011/02/04 17:47:27: SIP Rx udp:192.168.1.50:5060: INVITE sip:348xxxxxxx@172.16.10.210;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613 To: <sip:348xxxxxxx@172.16.10.210;user=phone> Call-ID: 240994b8@pbx CSeq: 7987 INVITE Max-Forwards: 70 Contact: <sip:123@192.168.1.50:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 P-Asserted-Identity: "vm" <sip:123@172.16.10.210> Content-Type: application/sdp Content-Length: 327 v=0 o=- 14816 14816 IN IP4 192.168.1.50 s=- c=IN IP4 192.168.1.50 t=0 0 m=audio 50782 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/02/04 17:47:27: Identify trunk (IP address/port and domain match) 12 [5] 2011/02/04 17:47:27: SIP Rx udp:192.168.1.50:5060: INVITE sip:348xxxxxxx@172.16.10.210;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613 To: <sip:348xxxxxxx@172.16.10.210;user=phone> Call-ID: 240994b8@pbx CSeq: 7987 INVITE Max-Forwards: 70 Contact: <sip:123@192.168.1.50:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 P-Asserted-Identity: "vm" <sip:123@172.16.10.210> Content-Type: application/sdp Content-Length: 327 v=0 o=- 14816 14816 IN IP4 192.168.1.50 s=- c=IN IP4 192.168.1.50 t=0 0 m=audio 50782 RTP/AVP 0 8 9 2 3 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtcp-xr:rcvr-rtt=all voip-metrics a=sendrecv [5] 2011/02/04 17:47:27: SIP Tx udp:192.168.1.50:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport=5060 From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613 To: <sip:348xxxxxxx@172.16.10.210;user=phone>;tag=d661df1270 Call-ID: 240994b8@pbx CSeq: 7987 INVITE Content-Length: 0 [5] 2011/02/04 17:47:27: Domain trunk pm@172.16.10.210 could not identify user for 348xxxxxxx [5] 2011/02/04 17:47:27: SIP Tx udp:192.168.1.50:5060: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport=5060 From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613 To: <sip:348xxxxxxx@172.16.10.210;user=phone>;tag=d661df1270 Call-ID: 240994b8@pbx CSeq: 7987 INVITE Contact: <sip:123@172.16.10.210:5060;transport=udp> Supported: 100rel, replaces, norefersub Allow-Events: refer Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE Accept: application/sdp User-Agent: snom-PBX/2011-4.2.0.3981 Content-Length: 0 [5] 2011/02/04 17:47:27: SIP Rx udp:192.168.1.50:5060: ACK sip:348xxxxxxx@172.16.10.210;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.50:5060;branch=z9hG4bK-30872cdc61a2b347876c6989f20f9f0f;rport From: "poa" <sip:300@192.168.1.50;user=phone>;tag=56613 To: <sip:348xxxxxxx@172.16.10.210;user=phone>;tag=d661df1270 Call-ID: 240994b8@pbx CSeq: 7987 ACK Max-Forwards: 70 Contact: <sip:123@192.168.1.50:5060;transport=udp> P-Asserted-Identity: "vm" <sip:123@172.16.10.210> Content-Length: 0 i can configure the sla or the blf of the remote extension ?? Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 4, 2011 Report Share Posted February 4, 2011 Okay, now you need to extract the destination number from the INVITE. You can see how this works on http://wiki.snomone.com/index.php?title=Inbounds_Calls#How_the_System_Routes_a_Call_to_the_Proper_Extension. Quote Link to comment Share on other sites More sharing options...
pbx support Posted February 4, 2011 Report Share Posted February 4, 2011 Make sure that your 3xx series extensions do not interfere with 348xxxxxxx numbers. If you want to go to PSTN (not to the other office), then make sure you have the proper dial plan setup. You can either the outside numbers with 1, so that you dial 1348xxxxxxx and the dial plan can have a pattern with 1*. Otherwise, you can have some complex pattern (regular expressions) for pattern matching to chose the right trunk. http://kiwi.pbxnsip.com/index.php/Dial_Plan Quote Link to comment Share on other sites More sharing options...
lirees Posted February 5, 2011 Author Report Share Posted February 5, 2011 I do not know if it's correct but i solved by changing the configuration of the both trunk in this way : Accept Redirect: yes Assume that call comes from user: 203 for office1 and 303 for office2 the extension 203 and 303 are a dummy user, in this way i can call form the office1 through the pstn and voip line of the office2 and viceversa now i should check with the blf of the snom320 in the office1 the status of the telephon in the office2, is it possible??? can i check also the sla ?? thanks Quote Link to comment Share on other sites More sharing options...
Vodia PBX Posted February 6, 2011 Report Share Posted February 6, 2011 now i should check with the blf of the snom320 in the office1 the status of the telephon in the office2, is it possible??? can i check also the sla ?? That would be a hack. You could set the snom 320 up in both offices (two identities) and use one identity just to monitor the status in the other domain. Quote Link to comment Share on other sites More sharing options...
lirees Posted February 8, 2011 Author Report Share Posted February 8, 2011 That would be a hack. You could set the snom 320 up in both offices (two identities) and use one identity just to monitor the status in the other domain. great !!! this is a fantastic workaround thank so much but Quote Link to comment Share on other sites More sharing options...
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